The present invention relates to loudspeakers. More particularly, the present invention relates to a loudspeaker, such as a subwoofer, which automatically calibrates itself when placed in a room to optimize an output signal of the loudspeaker for the room in which the loudspeaker is placed.
Designing speaker systems to produce high quality sound in home settings is a difficult task. Particularly, in the case of a subwoofer, the room in which the subwoofer is placed can cause standing waves or room modes which decrease sound quality.
More and more people are setting up high-end home theaters with at least one subwoofer as part of their system. These high-end systems are now approaching the performance of professional systems. When these high-end systems are put in a typical room, the room will often adversely affect the sound quality. Professional systems are usually installed in listening rooms that are carefully designed and which often use acoustic diffusers and sound-absorption material to improve the room acoustics. Most home users are un-likely to go to such length to improve their own home-theater or listening room. Either way, sound treatment of rooms with diffusers and absorption may still not produce a good acoustic room or it may only be optimal for just one position for the placement of speakers. Even in the most well designed room, standing waves exist that may make the low frequency response of the room un-even. The present invention electronically measures and quantifies these offending standing waves and reduces them to acceptable levels. The additional benefit of doing this is calibrating the room and having a known Sound Pressure Level (SPL). SPL measurements are made in decibels to reflect how loud a sound is perceived to be compared to the threshold of hearing.
The subwoofer of the illustrated embodiment, in addition to equalizing the room at low frequencies, has a number of other features. The illustrated subwoofer includes a USB and RS-232 control via a Personal Computer (PC) or home automation system, an advanced PC based GUI (Graphical User Interface), LCD display, SPL meter, firmware upgrades, remote control, diagnostic mode, demonstration mode, presets to store user preferences and settings in memory, tamper proof serial number, advanced limiter and is also capable of being connected to one or more subwoofers.
The following listed references are incorporated by reference herein. Throughout the specification, these references are referred to by citing to the numbers in the brackets [#] corresponding to each reference.                [1] Subwoofer Performance for Accurate Reproduction, Louis D. Fielder, Eric M. Benjamin, AES 83rd convention October 1987        [2] Output of a Sound Source in a Reverberation Chamber and Other Reflecting Environments, Richard V. Waterhouse, Journal of the Acoustical Society of America January 1958        [3] The Influence of Room Boundaries on Loudspeaker Power Output, Roy F. Allison, Journal of the Audio Engineers Society, June 1974        [4] An Exact Model of Acoustic Radiation in Enclosed Spaces, J. R. Wright, AES 96th convention February 1994        [5] Fundamentals of Acoustics, L. E. Kinsler, A. R. Frey, Wiley, New York, 1962        [6] High-Fidelity Sound System Equalization by Analysis of Standing Waves, Allen R. Groh, Journal of the Audio Engineers Society, June 1974        
Fielder and Benjamin in their paper [1] show that a 1 dB difference at low frequencies is just audible. Thus for accurate reproduction the subwoofer should be flat to within ±0.5 dB. They also state that room acoustics prevent the realization of such a goal. The embodiments disclosed herein provide results that approach this goal.
Waterhouse [2] plotted a room or boundary gain for a source with respect to distance. He showed that the boundary gain can be as much as 9 dB and is highest at the lowest frequencies with a slope of 12 dB per octave. As the sound source is moved away from the boundary, the gain remains the same but now occurs at a lower frequency. What is important to note, is that the slope at the lowest frequencies (12 per octave) remains the same. The situation is a little more complicated in a room as sound can be reflected back and forth repeatedly. However Waterhouse [2] shows his results are valid for rooms too. The results hold for any sized room, large or small. The size of the room is not relevant, just the distance from the walls is important. Allison [3] also presented 1974.
A subwoofer is usually positioned close to three boundaries (i.e. ⅛ space) as the ceiling of a room is acoustically too far away too make a difference to the frequency response. The room modes at the lowest frequencies are pretty sparse, see Table 1 from Wright [4].
TABLE 1Table of standing waves for a room of dimensions 4 m × 6 m × 2.5 mMode NumberFrequency HzMode Order(WLH)128.58010242.88100351.53110457.17020568.60001671.46120774.32011880.90101985.750301085.752001185.801111289.30021
Thus in the region of interest in calibrating a room, the slope we are interested in, is from 15 Hz to 25 Hz. For an average sized room of dimensions 4 m×6 m×2.5 m only one room exists near that band and it is at a frequency of 28.58 Hz. This means a subwoofer will produce the same signal between 15 Hz to 25 Hz in a normally constructed as in ⅛ space with only a gain difference between them.
Traditional methods of room equalization, both analog and digital have included ⅓-octave equalizers. To understand why this and other methods are inadequate consider a rectangular room with the dimension lx, ly and lz. Kinsler and Frey [5] developed the equation for the modes of the room as:
      F    xyz    =            c      2        ⁢                                        (                                          n                x                                            l                x                                      )                    2                +                              (                                          n                y                                            l                y                                      )                    2                +                              (                                          n                z                                            l                z                                      )                    2                    
Where nx, ny, nz=0, 1, 2, 3 . . .
And c is the speed of sound.
This equation's predicted room modes for the room of dimensions 4 m×6 m×2.5 m are listed in Table 1. The modes are very few at the lowest frequencies and progressively increase as the frequency goes up. Around 1 kHz the room modes have increased to a few thousand. The discrete number of room modes, only 12 at frequencies up to 90 Hz, show up as broad peaks and dips in the frequency response. The low frequency room modes bandwidth is dependent on the reverberation time. The lower the reverberation time, the larger the bandwidth, i.e. a room with very reflective walls and very little energy absorption at low frequencies will have very narrow room modes. Table 2, lists the relationship between modal bandwidth and reverberation time.
TABLE 2Table of Mode BandwithReverberation Time (s)Mode Bandwidth (Hz)0.2110.370.45.50.54.40.82.71.02.2
So, for a typical room, the long reverberation time makes the room modes more discrete. Q is related to the bandwidth with the following equation:
      N    =                  log        2            ⁡              (                              f            U                                f            C                          )                  Q    =                            2          N                                      2          N                -        1            
Where N is the bandwidth in octaves, fc is the center frequency of the mode, fu is the upper frequency at the −3 db point of the room mode. So, for example, with a reverberation time of 0.8 seconds, the mode bandwidth is 2.7 Hz. That means the lowest mode which is at 28.58 Hz is 0.07 octaves wide and has a Q of 20!! A ⅓ octave equalizer has a Q of 4.3. At higher frequencies of interest (like 70 Hz to 120 Hz) the discrete room modes will bunch together to produce a lower Q but this is totally dependent on the room dimension and the reverberation time of the room.
Groh [6] has shown that using pink noise to take a room response measurement will lead to an overly smoothed frequency response that will hide the peaky (high Q) nature of the room. If a chirp is used it must be long enough to get a good response of the room otherwise the measurement will be overly smoothed as with a pink noise measurement. Another technique is to use a MLS sequence but speaker non-linearity can corrupt the measurement.
According to an illustrated embodiment, a method of improving sound quality of a loudspeaker in a room is provided. The method includes providing a reference frequency response signal indicating a desired frequency response for the loudspeaker, measuring a frequency response of an output of the loudspeaker in the room, comparing the measured frequency response in the room to the reference frequency response signal, identifying at least one peak in the measured room frequency response which has a higher sound level than corresponding a sound level of the reference frequency response signal, and modifying the output of the loudspeaker to reduce the at least one peak identified in the identifying step without adjusting portions of the output of the loudspeaker having sound levels below corresponding sound levels of the reference frequency response signal.
Illustratively, the detecting step includes measuring a peak sound level generated in the room by the output of the loudspeaker at predetermined time intervals and storing the measured peak sound levels corresponding to different frequencies within the frequency range of the chirp sequence. Also illustratively, the method further includes converting the measured peak sound levels to sound pressure levels.
In another illustrated embodiment, the measuring step includes measuring a frequency response of the output signal in at least two different locations in the room and determining a combined measured frequency response based on the frequency response measurements taken in the at least two different locations in the room.
According to another illustrated embodiment, a method of improving sound quality of a loudspeaker in a room is provided. The method includes providing a reference frequency response signal indicating a desired frequency response for the loudspeaker, placing the loudspeaker in the room, initiating a chirp sequence over a predetermined frequency range for a predetermined time period greater than 10 seconds, detecting sound levels of an output of the loudspeaker at different frequencies within the frequency range during the chirp sequence, storing the detected sound levels to provide a measured frequency response of the output of the loudspeaker in the room, comparing the measured frequency response to the reference frequency response signal, and modifying the output of the loudspeaker based on the results of the comparing step.
In another example, the predetermined time period of the chirp sequence is greater than or equal to 48 seconds. In yet another example, the predetermined time period of the chirp sequence is greater than or equal to 55 seconds.
The chirp frequency range is illustratively from about 10 Hz to about 120 Hz for an subwoofer embodiment. Illustratively, the chirp sequence is generated at 1 Hz intervals within the frequency range, and a sound level of the output of the loudspeaker is detected and stored at each 1 Hz interval of the chirp sequence.
In one illustrated embodiment, the step of modifying the output of the loudspeaker uses frequency equalization. In other embodiments, the step of modifying the output of the loudspeaker uses at least one of output delay, phase change, or other signal processing technique.
According to yet another illustrated embodiment, a method of improving sound quality of a loudspeaker in a room is provided. The method includes providing a reference frequency response signal indicating a desired frequency response for the loudspeaker, measuring a frequency response of an output of the loudspeaker in the room, matching the reference frequency response signal with the measured frequency response by aligning the reference frequency response signal with the measured frequency response in a low frequency range, comparing the measured frequency response in the room to the reference frequency response signal after the matching step, and modifying the output of the loudspeaker based on the results of the comparing step.
In an illustrated example, the low frequency range for matching the reference frequency response signal with the measured frequency response is about 15 to about 25 Hz. In one example, the matching step is based on aligning a slope of the reference frequency response signal with a slope of the measured frequency response in the low frequency range. In another example, the matching step is based on aligning sound pressure levels of the reference frequency response signal with sound pressure levels of the measured frequency response in the low frequency range.
In yet another illustrated embodiment, the method further includes determining whether a difference between wherein the measured frequency response in the room and the reference frequency response signal exceeds a predetermined level after the matching step. The method also includes re-matching the reference frequency response signal with the measured frequency response if the difference exceeds the predetermined level.
According to another illustrated embodiment, a loudspeaker includes a housing, a speaker located in the housing, a digital signal processor located in the housing, and a memory located in the housing. The memory is coupled to the digital signal processor. The loudspeaker also includes an amplifier coupled to the digital signal processor, a speaker driver coupled to the amplifier and to the speaker, and a demonstration audio file stored in the memory. The digital signal processor is programmed to selectively retrieve the demonstration audio file and play it through the speaker without connecting the loudspeaker to a separate piece of audio equipment.
An illustrated embodiment also includes means for updating the demonstration audio file stored in the memory. Illustratively, the demonstration audio file is optimized for capabilities of the loudspeaker.
In another illustrated embodiment, the loudspeaker includes a user input device on the housing. The user input device is used to instruct the digital signal processor to retrieve the demonstration audio file and play it through the speaker. In yet another illustrated embodiment, a display is located on the housing. The display is coupled to the digital signal processor.
According to still another illustrated embodiment, a method is provided for demonstrating a loudspeaker. The method includes providing a speaker, a digital signal processor, a memory coupled to the digital signal processor, an amplifier, and a speaker driver coupled to the speaker within a housing, storing a demonstration audio file in the memory located within the housing, and executing a demonstration mode wherein the demonstration audio file is retrieved by the digital signal processor and played through the speaker using the amplifier and speaker driver in the housing without connecting the loudspeaker to external audio equipment.
In an illustrated embodiment, the method further includes compressing the demonstration audio file stored in the memory and decompressing the demonstration audio file for playback during the demonstration mode.
According to a further illustrated embodiment, a loudspeaker includes a housing, a speaker located within the housing, a controller located in the housing for driving the speaker; and a sound pressure level (SPL) detector located in the housing to measure a SPL of an output of the speaker.
In an illustrated embodiment, a display is located on the housing. The loudspeaker also includes means for displaying the measured SPL level detected by the SPL detector on the display. Illustratively, the means for displaying the measured SPL level also displays a frequency output of the speaker corresponding to the SPL level on the display.
According to another illustrated embodiment, a method includes providing a loudspeaker having a digital signal processor for controlling operation of the loudspeaker and a memory coupled to the digital signal processor, storing a unique serial number for the loudspeaker in the memory of the loudspeaker, and selectively retrieving the unique serial number from the memory.
In an illustrated embodiment, the method includes storing information related to the loudspeaker corresponding to the unique serial number, and retrieving the stored information based on the serial number retrieved from the memory. Illustratively, the stored information related to the loudspeaker includes at least one of a model number, a revision number, a date of manufacture, and a sales channel.
Also illustratively, the unique serial number is stored in a sector of a non-volatile memory during production of the loudspeaker. The sector is illustratively locked in software to reduce the likelihood of any change being made to the unique serial number. The sector may also be locked in hardware and made tamper-proof.
In another embodiment, the method further includes coupling a diagnostic tool to the digital signal processor of the loudspeaker and retrieving the unique serial number stored in the memory to facilitate at least one of maintenance, a repair, a recall, and an upgrade of the loudspeaker.
According to yet another illustrated embodiment, a method of operating a loudspeaker includes providing a loudspeaker having a digital signal processor for controlling operation of the loudspeaker and a memory coupled to the digital signal processor, storing a model number of the loudspeaker in the memory, and storing software in the memory for controlling the a plurality of different model numbers of loudspeakers. The method also includes determining the model number of the loudspeaker from the memory, selecting portions of software stored in the memory for controlling the loudspeaker based on the determined model number, and using the selected portions of the software to control the loudspeaker.
In an illustrated embodiment, the method further comprising storing information related to the loudspeaker corresponding to the model number, and retrieving the stored information based on the model number retrieved from the memory. In an other illustrated embodiment, the software determines appropriate filters to use to equalize an output of the loudspeaker based on the determined model number.
According to a further illustrated embodiment, a method of improving sound quality of a plurality of loudspeakers in a room includes providing a reference frequency response signal indicating a desired frequency response, measuring a combined frequency response of outputs from the plurality of loudspeakers in the room, and comparing the combined measured frequency response in the room to the reference frequency response signal. The method also includes modifying an output of a first loudspeaker based on the results of the comparing step, and using a modified output of the first loudspeaker as an input to at least one other loudspeaker.
Additional features of the present invention will become apparent to those skilled in the art upon consideration of the following detailed description of illustrative embodiments exemplifying the best mode of carrying out the invention as presently perceived.